Re: [Asterisk-Users] iconnect quality?

Jim Archer Tue, 11 Mar 2003 23:11:57 -0800

I pulled the CVS tree at 11:30PM (approximately) Eastern on March 11. This problem does seem to be fixed. But there is a new issue. It now seems that adding the 7777 prefix that solved the problems before now causes the "488 Not Acceptable Media" error response:

856.137213 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 856.283972 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 856.284294 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy Authentication Required 856.284651 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 856.285463 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol (proto=UDP 0x11, off=552) 856.285493 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE sip:[EMAIL PROTECTED] 856.423732 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 856.594346 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 856.598273 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 857.233615 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 858.123355 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 861.142317 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 867.170217 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media

Removing the 7777 prefix seems to make an improvement. The called phone will ring. If I answer it, though, Asterisk does not seem to know and keeps ringing my phone. If I call my cell phone and hang up Asterisk seems to notice, but if I call a POTS line, pick it up then hang up Asterisj seems not to notice:

56.265536 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 56.390918 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 56.391242 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy Authentication Required 56.391571 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 56.392423 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol (proto=UDP 0x11, off=552) 56.392453 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE sip:[EMAIL PROTECTED] 56.521701 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 59.769661 213.137.73.176 -> 192.203.175.9 SIP Status: 180 Ringing 64.637854 213.137.73.176 -> 192.203.175.9 SIP Status: 404 Not Found 78.543317 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 78.544346 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 80.062729 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 83.111677 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 88.659719 213.137.73.176 -> 192.203.175.9 SIP Status: 404 Not Found

Another example...

85.770587 192.203.175.9 -> 213.137.73.178 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description 85.897768 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 85.898134 213.137.73.176 -> 192.203.175.9 SIP Status: 407 Proxy Authentication Required 85.898581 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 85.899266 192.203.175.9 -> 213.137.73.178 IP Fragmented IP protocol (proto=UDP 0x11, off=552) 85.899294 192.203.175.9 -> 213.137.73.178 SIP Request: INVITE sip:[EMAIL PROTECTED] 86.097458 213.137.73.176 -> 192.203.175.9 SIP Status: 100 Trying 89.116528 213.137.73.176 -> 192.203.175.9 SIP Status: 180 Ringing 100.312843 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 100.313648 192.203.175.9 -> 213.137.73.178 SIP Request: ACK sip:[EMAIL PROTECTED] 101.812307 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 104.851267 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media 110.899191 213.137.73.176 -> 192.203.175.9 SIP Status: 488 Not Acceptable Media

--On Tuesday, March 11, 2003 8:04 PM -0600 Mark Spencer <[EMAIL PROTECTED]> wrote:

Actually I think it was an issue with incrementing the sequence number on
the bye andshould be fixed now.  RTCP is irrelevant in SIP signalling.

Mark

On Tue, 11 Mar 2003 [EMAIL PROTECTED] wrote:

> 1 - From watching the udp fly by, it seems that iconnect does not know
> when we hang up.  For example, if I call a voice mail and it starts
> giving me its speal and I hang up, iconnect stays connected until the
> VM hangs up at its end.
Because Asterisk doesn't implement RTCP.


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